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Side by Side Diff: webrtc/voice_engine/test/auto_test/fakes/conference_transport.h

Issue 1652983002: Remove mutable from rtc::CriticalSections. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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121 121
122 static bool Run(void* transport) { 122 static bool Run(void* transport) {
123 return static_cast<ConferenceTransport*>(transport)->DispatchPackets(); 123 return static_cast<ConferenceTransport*>(transport)->DispatchPackets();
124 } 124 }
125 125
126 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const; 126 int GetReceiverChannelForSsrc(unsigned int sender_ssrc) const;
127 void StorePacket(Packet::Type type, const void* data, size_t len); 127 void StorePacket(Packet::Type type, const void* data, size_t len);
128 void SendPacket(const Packet& packet); 128 void SendPacket(const Packet& packet);
129 bool DispatchPackets(); 129 bool DispatchPackets();
130 130
131 mutable rtc::CriticalSection pq_crit_; 131 rtc::CriticalSection pq_crit_;
132 mutable rtc::CriticalSection stream_crit_; 132 rtc::CriticalSection stream_crit_;
133 const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_; 133 const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
134 rtc::PlatformThread thread_; 134 rtc::PlatformThread thread_;
135 135
136 unsigned int rtt_ms_; 136 unsigned int rtt_ms_;
137 unsigned int stream_count_; 137 unsigned int stream_count_;
138 138
139 std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_); 139 std::map<unsigned int, std::pair<int, int>> streams_ GUARDED_BY(stream_crit_);
140 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_); 140 std::deque<Packet> packet_queue_ GUARDED_BY(pq_crit_);
141 141
142 int local_sender_; // Channel Id of local sender 142 int local_sender_; // Channel Id of local sender
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154 webrtc::VoENetwork* remote_network_; 154 webrtc::VoENetwork* remote_network_;
155 webrtc::VoEFile* remote_file_; 155 webrtc::VoEFile* remote_file_;
156 156
157 LoudestFilter loudest_filter_; 157 LoudestFilter loudest_filter_;
158 158
159 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_; 159 const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
160 }; 160 };
161 } // namespace voetest 161 } // namespace voetest
162 162
163 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_ 163 #endif // WEBRTC_VOICE_ENGINE_TEST_AUTO_TEST_FAKES_CONFERENCE_TRANSPORT_H_
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